Low/High pass filters and sampling rates are totally different. "MOST" human ears cant hear FREQUENCIES past 18.5 kHz – 20 kHz, and that is what a cutoff's purpose is - to filter out the high frequencies a person cant hear and would otherwise waste space. SAMPLEING RATES refer to how many times a sound card takes a "snapshot" of the sound - think of it the same way you think of movie frame rates. CD sound has a sampling rate of 44.1 kHz, and DVD audio has a 48 kHz sampling rate. Going below a 44.1 kHz sampling rate greatly hurts sound quality - its like listening to a radio. Just record a sound at a 22.050 kHz sampling rate and see what I mean. Also, the quality re-sampling a CD's 44.1 kHz sampling rate to 40 kHz would further degrade sound quality. Most sound cards probably don't support irregular sampling rates either, so a 40 kHz sampling rate is a moot point. Hope I’ve helped you out a bit.

In the early days of digital audio research, the necessary bandwidth of about 1 Mbps per audio channel was difficult to store. Disk drives had the bandwidth but not the capacity for long recording time, so attention turned to video recorders. These were adapted to store audio samples by creating a pseudo-video waveform which would convey binary as black and white levels. The sampling rate of such a system is constrained to relate simply to the field rate and field structure of the television standard used, so that an integer number of samples can be stored on each usable TV line in the field. Such a recording can be made on a monochrome recorder, and these recording are made in two standards, 525 lines at 60 Hz and 625 lines at 50 Hz. Thus it is possible to find a frequency which is a common multiple of the two and is also suitable for use as a sampling rate.

The allowable sampling rates in a pseudo-video system can be deduced by multiplying the field rate by the number of active lines in a field (blanking lines cannot be used) and again by the number of samples in a line. By careful choice of parameters it is possible to use either 525/60 or 625/50 video with a sampling rate of 44.1KHz.

In 60 Hz video, there are 35 blanked lines, leaving 490 lines per frame or 245 lines per field, so the sampling rate is given by :

60 X 245 X 3 = 44.1 KHz

In 50 Hz video, there are 37 lines of blanking, leaving 588 active lines per frame, or 294 per field, so the same sampling rate is given by

50 X 294 X3 = 44.1 Khz.

The sampling rate of 44.1 KHz came to be that of the Compact Disc. Even though CD has no video circuitry, the equipment used to make CD masters is video based and determines the sampling rate.

The 44.1 Khz sample rate is used in Mp3s and such because most of them are ripped from CDs and resampling is was not as good as it is today (Not too good still)..

BTW

QUOTE

Sampling vs Frequency Response: The sampling frequency must be at least twice as high as the highest frequency that you wish to reproduce because you must have at least 1 data point for each half cycle of the audio waveform. The highest frequency that you can record with a sampling rate of 8k is 4000hz. At a sampling rate of 44k, you can record up to 22khz but the filters used in the D/A conversion process have a very high rate slope at 20,000hz which will allow nothing higher than 20khz to get through. The newer D/A converters with high rates of oversampling are able to use low pass filters with a slower roll off.

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"You have the right to remain silent. Anything you say will be misquoted, then used against you."

The 44.1 Khz sample rate is used in Mp3s and such because most of them are ripped from CDs and resampling is was not as good as it is today (Not too good still)..

Fascinating! Thanks for the quote.

I remember Frank Klemm mentioned all this a while back (here's the post, which contains some other neat tidbits), but at the time I couldn't quite understand what he meant (a common problem when I try to communicate with electrical engineers ).

If you can or can not hear above 20Khz etc you can percieve it on at least a subconcious level. Sustained high frequency sound combined with psychotropic compunds etc has been show to have a definate impact on the human nervous system IIRC. Often invoking unprovoked anger or rage that would not otherwise manifest.

If you can or can not hear above 20Khz etc you can percieve it on at least a subconcious level. Sustained high frequency sound combined with psychotropic compunds etc has been show to have a definate impact on the human nervous system IIRC. Often invoking unprovoked anger or rage that would not otherwise manifest.

Maybe we can just make a new audio format that will also reach out from the stereo and repeatedly poke you with a stick >_<. That would invoke some anger and rage just as effectively as high-frequency reproduction, if not moreso.

So why don't just use a 40 khz PCM instead of 44.1 khz as human can't listen to frequency above 20 khz?

One of my freshman university courses said it's an "engineering-margin". You take a little more than 2 x 20 kHz just in case, the system will always have flaws and imperfections, so they'd rather not live 'on the edge'. So for example if one of your filters cuts off a few hundred Hz too soon, it's still unhearable.

Why include high frequency noises if you can't hear them and they only give you headaches and make you angry?

So why don't just use a 40 khz PCM instead of 44.1 khz as human can't listen to frequency above 20 khz?

IMO That's because some people can hear even more than 20kHz....up to 22kHz.

Me, for example, with my equipment (SBLive 5.1, FPS1000 4.1 speakers) I've generated sine waves (sampling rate 48000Hz, of course.....) and I can hear up to 20kHz (the speakers' and soundcard' response is no more than 20kHz, if more juice is used then downpitching occurs)Of course, any frequency above 16kHz, for me, requires additional amplification (normalization) gradually....but I can actually hear 'em. Would be nice if I could test for hearing above 20kHz

In the digital domain, yes. (example, generating tones in a wav editor).But the digital side effects (infinite ringing) can add distortion in the analog domain, to avoid this, the DACs and ADCs used have a frequency response slightly inferior.